Wireless communication networks are widely deployed to provide various communication services such as telephony, video, data, messaging, broadcasts, video telephony, and so on. Such networks, which are usually multiple access networks, support communications for multiple users by sharing the available network resources. One example of such a network is the UMTS Terrestrial Radio Access Network (UTRAN). The UTRAN is the radio access network (RAN) defined as a part of the Universal Mobile Telecommunications System (UMTS), a third generation (3G) mobile phone technology supported by the 3rd Generation Partnership Project (3GPP). UMTS, which is the successor to Global System for Mobile Communications (GSM) technologies, currently supports various air interface standards, such as Wideband-Code Division Multiple Access (W-CDMA), Time Division-Code Division Multiple Access (TD-CDMA), and Time Division-Synchronous Code Division Multiple Access (TD-SCDMA). UMTS also supports enhanced 3G data communications protocols, such as High Speed Packet Access (HSPA), which provides higher data transfer speeds and capacity to associated UMTS networks.
As the demand for mobile broadband access continues to increase, research and development continue to advance wireless communication technologies (e.g., UMTS technologies) not only to meet the growing demand for mobile broadband access, but to advance and enhance the user experience with mobile communications such as improved video telephony.
Due to tight delay constraints, video telephony (VT) applications typically track the bandwidth available in a network to ensure that data rates of the VT applications do not exceed the available bandwidth. VT applications also may monitor the delay from one end of the call to the other (e.g., from one mobile terminal to another mobile terminal). By tracking and controlling the data rate of the VT application, occurrence of disturbances such as frame freezes, drops, and blurs may be reduced. In the related art, VT applications that operate over internet protocol (IP) typically monitor various network metrics such as bandwidth, round-trip time (RTT), jitter at the transport layer (e.g., TCP/UDP) or higher layer, etc. A VT application running at a device relies on the other end of the video call to inform the VT application of the network metrics being monitored. This end-to-end feedback mechanism arrangement introduces an undesirable reaction delay when congestion is detected in a data packet path, especially when the congestion is on the uplink of the sender.
Furthermore, end-to-end feedback mechanisms generally do not indicate how much more bandwidth is available for the video telephony call. When congestion occurs, the data path between the sender and receiver does not provide sufficient uplink and/or downlink bandwidth for transmitting data without causing undesirable effects such as queuing delay and/or packet loss. Lack of congestion simply indicates that bandwidth is available, but no useful information is provided as to how much more bandwidth is available. In wireless communication, tracking the available bandwidth of wireless channels is even more challenging because the available wireless bandwidth is often time-varying. Therefore, due to the lack of available bandwidth information, end-to-end feedback-based rate adaptation algorithms for video communication are unable to aggressively utilize the available bandwidth.